High-speed display of 3-D objects in virtual reality environments is one of the currently important subjects. Shape simplification is considered an efficient method. This paper presents a method of hierarchical cube-based segmentation for shape simplification and multiresolution model construction. The relations among shape simplification, resolution and visual distance are derived firstly. The first level model is generated from scattered range data by cube-base segmentation with the first level cube size. Multiresolution models are then generated by re-sampling polygonal patch vertices of each former level model with hierarchical cube-based segmentation structure. The results show that the algorithm is efficient for constructing multiresolution models of free-form shape 3-D objects from scattered range data and high compression ratio can be obtained with little noticeable difference during the visualization.

Electroluminescent (EL) devices with mixed single layer that consist of fluorescent dyes, distylylbiphenyl derivative (DPVBi) and triphenylamine derivative (TPD), are studied. Blue light emission was observed from the device with DPVBi and TPD. White emission over 2,500 cd/m2 was observed from the devices with mixed single layer of DPVBi, TPD and dicyanomethylene derivative (DCM).

This paper proposes a nonlinear signal processing by using a three layered network which is trained with self-organized clustering and supervised learning. The network consists of three layers, i.e., self-organized layer, an evaluation layer and an output layer. Since the evaluation layer is designed as a simple perceptron network and the output layer is designed as a fixed weight linear node, the training complexity is the same as a conventional one consisting of self-organized clustering and a simple perceptron network. In other words, quite high speed training can be realized. Generally speaking, since the data range is arbitrary large in signal procession, the network shoulk cover this range and output a value as accurately as possible. However, it may be hard for only a node in the network to output these data. Instead of this mechanism, if this dynamic range is covered by using several nodes, the complexity of each node is reduced and the associated range is also limited. This results on the higher performance of the network than conventional RBFs. This paper introduces a new non-linear spectrum estimation which consists of LPC analysis and RBF network. It is shown that accuracy spectrum envelopes can be obtained since a new RBF network can estimate some nonlinearities in a speech production.

An extension is made for a set of systems that have a quadratic Lyapunov function in common for the purpose of analysis and design. The nominal set of system matrices comprises stable symmetric matricies, which admit a hyperspherical Lyapunov function. Based on stability robustness results, sets of matrices are constructed so that they share the same Lyapunov function with the nominal ones.

Maximal-ratio combining (MRC), which maximizes the carrier to noise ratio (CNR) of the combined signal, generally requires envelope detection and multiplication having linear characteristic over a wide dynamic range to generate a weighting factor for each branch. In this paper, we propose a simplified two-branch diversity combining scheme without linear envelope detection. The proposed scheme, called "level comparison weighted combining (LCWC),"is simplified in a manner that its weighting factor for each branch is generated from hard-decision results of comparing signal envelopes between two branches. Performance of LCWC is evaluated by computer simulation and laboratory experiment, which shows that its diversity gain is almost identical to that of MRC in a Rayleigh fading channel.

This paper presents a clock routing technique called Balanced-Mesh Method (BMM) which incorporates the advantages of two famous conventional-clock-routing techniques. One is the balanced-tree method (BTM) where the clock net is routed as a tree so that the delay times of clock signal are balanced, and the other is the fixed-mesh method (FMM) where the clock net is routed as a fixed mesh driven by a large buffer. In BMM, the clock net is routed as a set of relatively small meshes of interconnects driven by relatively small buffers. Each mesh covers an area called a Mesh-Routing Region (MR) in which its delay and skew can be suppressed within a certain range. These small meshes are connected by a balanced tree with the chip clock source as its root. To implement BMM, we developed an MR-partitioning program that partitions the circuit into MR's according to a set of pre-determined constraints on the number of flip-flops and the area in each MR, and a clock-global-routing program that provides each mesh routing and the tree routing connecting meshes. We applied BMM to the design of an MPEG2-encoder LSI and achieved a skew of 210ps. In addition, the experimental results show BMM yields the lowest power dissipation compared to conventional methods.

Many network protocols for routing messages have been proposed for mobile computing environments. In this paper, we consider the query processing strategy which operates over these network protocols. To begin with, we introduce five fundamental location update methods based on ideas extracted from the representative network protocols. They are the single broadcast notification (SBN), the double broadcast notification (WBN), the single default notification (SDN), the double default notification (WDN), and the no notification (NN). As a network protocol, each method is strong in performance in some system enrivonment, but weak in others. In practical situations, where various kinds of applications are used for various purposes, however, it is required to use a single method. We therefore propose an adaptive query processing strategy where these five location update methods can be dynamically selected. Moreover, we analyze the performance of this adaptive query processing strategy via the Markov chain. We also use the statistical approach to estimate the traffic of individual hosts. Finally, we show the efficiency of our proposed strategy over a wide area of system environments.

This paper introduces a new recursive factorization of the polynomial, 1-zN, over the real numbers when N is an even composite integer. The recursive factorization is applied for efficient computation of the discrete Fourier transform (DFT) and the cyclic convolution of real sequences with highly composite even length.

An analytic traffic model is presented to estimate the soft handoff rate in DS-CDMA cellular systems. The model is based on the fact that a mobile in soft handoff call is connected to two cell sites when it is in an overlapped region. The handoff rate is estimated by the mobility of mobiles, which is a function of the size and shape of cell area, and the call density and speed of mobiles in the area. Simulation results show good agreement with the analytical model.

Block delay caused by using fast Fourier transform (FFT), and computational complexity in sampling rate conversion system are considered in this paper. The relationship between the number of block delays and the computational complexity is investigated. The proposed method can avoid the redundant operations of sampling rate conversion completely and moreover provide a good trade-off between the number of block delays and the computational complexity. As a result, ti is shown that with the proposed method, the sampling rate conversion can be realized more efficiently under a small number of block delays.

Within the framework of a previously proposed vision system, a new part-segmentation algorithm, that breaks an object defined by its contour into its constituent parts, is presented. The contour is assumed to be obtained using an edge detector. This decomposition is achieved in two stages. The first stage is a preprocessing step which consists of extracting the convex dominant points (CDPs) of the contour. For this aim, we present a new technique which relaxes the compromise that exists in most classical methods for the selection of the width of the Gaussian filter. In the subsequent stage, the extracted CDPs are used to break the object into convex parts. This is performed as follows: among all the points of the contour only the CDPs are moved along their normals nutil they touch another moving CDP or a point on the contour. The results show that this part-segmentation algorithm is invariant to transformations such as rotation, scaling and shift in position of the object, which is very important for object recognition. The algorithm has been tested on many object contours, with and without noise and the advantages of the algorithm are listed in this paper. Our results are visually similar to a human intuitive decomposition of objects into their parts.

A noise reduction filter composed of a sandglass-type neural network (Sandglass-type Neural network Noise Reduction Filter: SNNRF) was proposed in the present paper. Sandglass-type neural network (SNN) has symmetrical layer construction, and consists of the same number of units in input and output layers and less number of units in a hidden layer. It is known that SNN has the property of processing signals which is equivalent to KL expansion after learning. We applied the recursive least square (RLS) method to learning of SNNRF, so that the SNNRF became able to process on-line noise reduction. This paper showed theoretically that SNNRF behaves most optimally when the number of units in the hidden layer is equal to the rank of covariance matrix of signal component included in input signal. Computer experiments confirmed that SNNRF acquired appropriate characteristics for noise reduction from input signals, and remarkably improved the SN ratio of the signals.

In this paper, a new trial for the signal processing is proposed along the same line as a previous study on the extended regression analysis based on the Bayes' theorem. This method enables us to estimate a response probability property of complicated systems in an actual case when observation values of the output response are roughly observed due to the quantization mechanism of measuring equipment. More concretely, the main purpose of this research is to find the statistics of the joint probability density function before a level quantization operation which reflects every proper correlation informations between the system input and the output fluctuations. Then, the output probability distribution for another kind of input is predicted by using the estimated regression relationship. Finally, the effectiveness of the proposed method is experimentally confirmed by applying it to the actually observed input-output data of the acoustic system.

Introducing a mathematical model of noise in stereo images, we propose a new criterion for intelligent statistical inference about the scene we are viewing by using the geometric information criterion (geometric AIC). Using synthetic and real-image experiments, we demonstrate that a robot can test whether or not the object is located very far away or the object is a planar surface without using any knowledge about the noise magnitude or any empirically adjustable thresholds.

This paper describes a method to distinguish phase noise and amplitude noise from total oscillator noise in circuit simulation, and derives general relationships between periodic time-varying transfer functions for oscillators and phase and amplitude noises.

This paper describes the performance of a video communication system over mobile radio channels. Mobile channel quality changes rapidly due to various factors. When compressed video data is transmitted through these channels, it is indispensable to employ an error control scheme because reconstructed video quality is seriously degraded by channel error. To control this error, an automatic repeat request (ARQ) scheme is often employed, however, this incurs a cost. The benefit of a non-degraded reconstructed video sequence is offset by the transmission delay due to ARQ retransmission. We apply to a video communication system a selective-repeat ARQ which is combined with the coding control scheme to reduce the transmission delay. We evaluate the quality of the reconstructed video sequence and transmission delay using computer simulations and make clear its applicability over Rayleigh and Nakagami-Rican fading channels and intersymbol interference.

It is indispensable to establish a multi-access protocol and resource management technique that can assure transmission quality and efficiently utilize the radio frequency spectrum for ATM-based wireless access systems. This paper proposes dynamic time-slot assignment schemes for the forward link from a user to a central station (CS): (1) the centralized assignment and release scheme (CAR), and (2) the centralized-assignment and autonomous-release scheme (CAAR). In the proposed schemes, a central station dynamically assigns time-slots based on traffic information obtained by monitoring the input traffic in each radio module (RM). In addition, forward protection is used to prevent false-release of assigned time-slots. Performance evaluations have been carried out by analysis as well as computer simulations. They show that the proposed schemes achieve good performance in delay, link stability, and utilization efficiency of radio resources with an optimized number of forward protection steps.

This paper proposes a data transfer protocol called Logical Airlink Control Procedure for Packet Radio Systems (LAPPR) which offers a high throughput efficiency in fading channels. Two main ideas are presented in the flow control and retransmission control methods. A new flow control scheme called "P/F Sliding Flow" is proposed where a high channel utilization rate is provided by continuously transmitting the data frames. At that time, the acknowledgements for the data frames are reduced so as not to occupy both the forward and reverse channels and data exchange between different transmitter-receiver pairs is enabled. A retransmission control scheme called "Recovery Confirming Retransmission Control" is also proposed which makes conventional Automatic Repeat Request (ARQ) schemes significantly more effective in the fading environments by continuously sending the same frame. Computer simulations were carried out and the result showed that a throughput higher than conventional protocols can be achieved.

In this paper, a systematic method which synthesizes the datapath of Application Specific Instruction Processor (ASIP) is proposed. The behavioral description of application is written in instruction code defined on abstract machine. We introduce register transfer graph (RTG) to represent instructions and synthesis constraint tree to select the combinations of synthesis constraints to explore design space along area and performance axis. The high performance is achieved by scheduling micro-operations of instruction in out-of-order. The practical datapath is synthesized by considering connection geometry as well as the maximum utilization of hardware resources. To reduce connection cost, data transfer paths are minimized by replacing an inefficient data transfer path with its bypass route. The feasibility of the proposed synthesis method is verified with several experimental instruction sequences.

We propose a soft decision Viterbi decoding scheme and a self-interference cancellation method applicable to a Parallel Combinatory CDMA (PC-CDMA) system. In this decoding scheme, branch metric is calculated for every bit by weighting the output levels of the PC-CDMA correlators so as to enable an effective soft decision capability to the system. The effectivity of this scheme is then further enhanced by the use of a simple pseudo-random bit interleaving scheme. Moreover, to increase the capacity of the PC-CDMA system, we propose a simple self-interference cancellation method for self-induced cross-correlation arising from the multipath environment. This further enhances the efficacy of the decoding scheme because the false contributions of the self-induced cross-correlation component are removed from the branch metric prior to soft decision Viterbi decoding. Finally, we simulated a possible PC-CDMA system with a user data rate of 1.92Mbps, transmitting it at a chip rate of 3.84Mcps and at 7.68Mcps under a multipath-Rayleigh fading interference environment. For a chip rate of 7.68Mcps, BER after Viterbi decoding is less than 3.2e-7 even without the use of interference cancellation. For a chip rate of 3.84Mcps, BER after Viterbi decoding with interference cancellation is 1.0e-4.